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Quisk version 4.2.45

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This commit is contained in:
James C. Ahlstrom
2025-10-08 16:29:31 -04:00
parent 458e43252d
commit 0528e8bbbe
17 changed files with 97 additions and 1331 deletions
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quisk.c
+8 -123
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@@ -151,7 +151,6 @@ static int waterfall_scroll_mode = 1; // draw the first lines multiple times
static int quisk_use_sidetone; // is there a sidetone volume control?
static int hl2_txbuf_errors; // errors in the Hermes-Lite2 Tx buffer
static int hl2_txbuf_state; // state machine for errors in the Hermes-Lite2 Tx buffer
static int freedv_monitor; // pass the Freedv audio to the speakers instead of to the demodulator
static int softrock_correct_freq; // gain and phase corrections for SoftRock
static double softrock_correct_gain;
static double softrock_correct_phase;
@@ -569,15 +568,10 @@ void QuiskWavClose(struct QuiskWav * hWav)
}
#endif
// These are used for digital voice codecs
ty_dvoice_codec_rx pt_quisk_freedv_rx;
ty_dvoice_codec_tx pt_quisk_freedv_tx;
void quisk_dvoice_freedv(ty_dvoice_codec_rx rx, ty_dvoice_codec_tx tx)
void quisk_dvoice_freedv(void) // Not used.
{
pt_quisk_freedv_rx = rx;
pt_quisk_freedv_tx = tx;
}
#if 0
static int fFracDecim(double * dSamples, int nSamples, double fdecim)
{ // fractional decimation by fdecim > 1.0
@@ -1851,10 +1845,8 @@ static int quisk_process_decimate(complex double * cSamples, int nSamples, int b
static int quisk_process_demodulate(complex double * cSamples, double * dsamples, int nSamples, int bank, int nFilter, rx_mode_type rx_mode)
{ // Changes here will require changes to get_filter_rate();
int i;
bool isLSB;
complex double cx;
double d, di, dd;
static struct AgcState Agc1 = {0.3, 16000, 0}, Agc2 = {0.3, 16000, 0};
//static int count=0;
//static double phase=0;
static struct stStorage {
@@ -2090,6 +2082,7 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
MeasureSquelch[bank].squelch_active = MeasureSquelch[bank].squelch < squelch_level;
break;
case DGT_U: // digital mode DGT-U at 48 ksps
case FDV_U:
if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) { // filter at 6 ksps
quisk_filter_srate = quisk_decim_srate / 8;
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
@@ -2116,6 +2109,7 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
}
break;
case DGT_L: // digital mode DGT-L
case FDV_L:
if (filter_bandwidth[nFilter] < DGT_NARROW_FREQ) { // filter at 6 ksps
quisk_filter_srate = quisk_decim_srate / 8;
nSamples = quisk_cDecim2HB45(cSamples, nSamples, &Storage[bank].HalfBand5);
@@ -2154,102 +2148,6 @@ static int quisk_process_demodulate(complex double * cSamples, double * dsamples
}
}
break;
case FDV_U: // digital voice
case FDV_L: // Extra modes added by Dave Roberts, G8KBB, June 2020. Modified by N2ADR 2024.
quisk_check_freedv_mode();
// current coding assumes input rate n_modem_sample_rate is 8000 or 48000.
// decimate:
switch (n_modem_sample_rate) {
case 8000:
quisk_filter_srate = quisk_decim_srate / 6;
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim48to16, 3);
nSamples = quisk_cDecimate(cSamples, nSamples, &Storage[bank].filtDecim16to8, 2);
break;
case 48000:
quisk_filter_srate = quisk_decim_srate;
break;
default:
quisk_filter_srate = quisk_decim_srate / 6;
QuiskPrintf("Bad FreeDV modem rate in quisk_process_demodulate\n");
break;
}
// filter and demodulate to mono audio:
isLSB = (rx_mode == FDV_L);
complex double cBuf[nSamples];
for (i = 0; i < nSamples; i++) {
cx = cRxFilterOut(cSamples[i], bank, nFilter);
if (isLSB)
dd = creal(cx) + cimag(cx);
else
dd = creal(cx) - cimag(cx);
if(bank == 0) {
measure_audio_sum += dd * dd;
measure_audio_count += 1;
}
cBuf[i] = dd;
}
// Perhaps decimate by an additional fraction
if (quisk_decim_srate != 48000) {
dd = quisk_decim_srate / 48000.0;
nSamples = cFracDecim(cBuf, nSamples, dd);
quisk_decim_srate = 48000;
}
if (bank == 0)
process_agc(&Agc1, cBuf, nSamples, 0);
else
process_agc(&Agc2, cBuf, nSamples, 0);
// pass data decimated down to n_modem_sample_rate
// return data is at n_speech_sample_rate
#if 0
static double dmax_in = 0;
for (i = 0; i < nSamples; i++) {
dd = fabs(creal(cBuf[i])) / CLIP16;
if (dd > dmax_in)
dmax_in = dd;
}
if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
//printf("Max sample in %.0lf\n", dmax_in);
dmax_in = 0;
}
#endif
if (nSamples > n_max_modem_samples) {
printf ("FreeDV: nSamples %d exceeds max_modem_samples %d\n", nSamples, n_max_modem_samples);
nSamples = n_max_modem_samples;
}
short sBuf[nSamples];
for (i = 0; i < nSamples; i++)
sBuf[i] = (short)(creal(cBuf[i]) / CLIP16);
if (freedv_monitor) {
for (i = 0; i < nSamples; i++)
dsamples[i] = (float)sBuf[i] * CLIP16;
}
else {
if (pt_quisk_freedv_rx)
nSamples = (* pt_quisk_freedv_rx)(sBuf, dsamples, nSamples, bank);
}
#if 0
static double dmax_out = 0;
for (i = 0; i < nSamples; i++) {
dd = fabs(dsamples[i]) / CLIP16;
if (dd > dmax_out)
dmax_out = dd;
}
if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
//printf(" Max sample out %.0lf\n", dmax_out);
dmax_out = 0;
}
#endif
// current coding assumes output rate n_speech_sample_rate is 8000 or 16000
switch (n_speech_sample_rate) {
case 8000:
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 3);
nSamples = quisk_dInterp2HB45(dsamples, nSamples, &Storage[bank].HalfBand7);
break;
case 16000:
nSamples = quisk_dInterpolate(dsamples, nSamples, &Storage[bank].filtAudio24p3, 3);
break;
}
break;
}
if (bank == 0 && measure_audio_count >= quisk_filter_srate * measure_audio_time) {
measured_audio = sqrt(measure_audio_sum / measure_audio_count) / CLIP32 * 1e6;
@@ -2785,9 +2683,6 @@ start_agc:
if (rxMode == EXT || rxMode == DGT_IQ) { // Ext and DGT-IQ stereo sound
process_agc(&Agc1, cSamples, nSamples, 1);
}
else if (rxMode == FDV_U || rxMode == FDV_L) { // Agc already done
;
}
else if (split_rxtx || multirx_play_channel >= 0) { // separate AGC for left and right channels
for (i = 0; i < nSamples; i++) {
orig_cSamples[i] = cimag(cSamples[i]);
@@ -2945,6 +2840,8 @@ static PyObject * get_filter_rate(PyObject * self, PyObject * args)
break;
case DGT_U: // digital modes DGT-*
case DGT_L:
case FDV_U: // digital voice
case FDV_L:
if (bandwidth < DGT_NARROW_FREQ)
filter_srate = decim_srate / 8;
else
@@ -2953,10 +2850,6 @@ static PyObject * get_filter_rate(PyObject * self, PyObject * args)
case DGT_IQ: // digital mode at 48 ksps
filter_srate = decim_srate;
break;
case FDV_U: // digital voice
case FDV_L:
filter_srate = n_modem_sample_rate;
break;
}
//QuiskPrintf("Filter rate %d\n", filter_srate);
return PyInt_FromLong(filter_srate);
@@ -3196,14 +3089,13 @@ static PyObject * set_params(PyObject * self, PyObject * args, PyObject * keywds
{ /* Call with keyword arguments ONLY; change local parameters */
static char * kwlist[] = {"quisk_is_vna", "rx_bytes", "rx_endian", "read_error", "clip",
"bscope_bytes", "bscope_endian", "bscope_size", "bandscopeScale", "hermes_pause",
"freedv_monitor", NULL} ;
NULL} ;
int i, nbytes, read_error, clip, bscope_size, hermes_pause;
nbytes = read_error = clip = bscope_size = hermes_pause = -1;
if (!PyArg_ParseTupleAndKeywords (args, keywds, "|iiiiiiiidii", kwlist,
&quisk_is_vna, &nbytes, &py_sample_rx_endian, &read_error, &clip,
&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause,
&freedv_monitor))
&py_bscope_bytes, &py_bscope_endian, &bscope_size, &bandscopeScale, &hermes_pause))
return NULL;
if (nbytes != -1) {
py_sample_rx_bytes = nbytes;
@@ -6269,13 +6161,6 @@ static PyMethodDef QuiskMethods[] = {
{"set_cwkey", set_hardware_cwkey, METH_VARARGS, "Change the CW key up/down state."},
{"set_remote_cwkey", set_remote_cwkey, METH_VARARGS, "Change the remote control CW key up/down state."},
{"set_PTT", set_PTT, METH_VARARGS, "Record the PTT button state."},
{"freedv_open", quisk_freedv_open, METH_VARARGS, "Open FreeDV."},
{"freedv_close", quisk_freedv_close, METH_VARARGS, "Close FreeDV."},
{"freedv_get_snr", quisk_freedv_get_snr, METH_VARARGS, "Return the signal to noise ratio in dB."},
{"freedv_get_version", quisk_freedv_get_version, METH_VARARGS, "Return the codec2 API version."},
{"freedv_get_rx_char", quisk_freedv_get_rx_char, METH_VARARGS, "Get text characters received from freedv."},
{"freedv_set_options", (PyCFunction)quisk_freedv_set_options, METH_VARARGS|METH_KEYWORDS, "Set the freedv parameters."},
{"freedv_set_squelch_en", quisk_freedv_set_squelch_en, METH_VARARGS, "Enable or disable FreeDV squelch."},
{"wdsp_set_parameter", (PyCFunction)quisk_wdsp_set_parameter, METH_VARARGS|METH_KEYWORDS, "Set parameters for the WDSP SDR library."},
{"tmp_record_save", tmp_record_save, METH_VARARGS, "Save the temporary recording in a WAV file."},
{"watfall_RgbData", watfall_RgbData, METH_VARARGS, "Return a cookie for the Waterfall pixel data."},